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we ne allow audioOutput plugins to set the final outAudioFormat that will be used. we now use alsa's _near functions to detect what to use.

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git-svn-id: https://svn.musicpd.org/mpd/trunk@3038 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes
2005-03-08 01:53:13 +00:00
parent fa08a7e3ff
commit b5e55a648f
3 changed files with 27 additions and 13 deletions
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25
src/audioOutput.c
View File

@@ -73,13 +73,14 @@ AudioOutput * newAudioOutput(ConfigParam * param) {
memset(&ret->inAudioFormat, 0, sizeof(AudioFormat)); memset(&ret->inAudioFormat, 0, sizeof(AudioFormat));
memset(&ret->outAudioFormat, 0, sizeof(AudioFormat)); memset(&ret->outAudioFormat, 0, sizeof(AudioFormat));
memset(&ret->reqAudioFormat, 0, sizeof(AudioFormat));
getBlockParam(AUDIO_OUTPUT_FORMAT, format, 0); getBlockParam(AUDIO_OUTPUT_FORMAT, format, 0);
if(format) { if(format) {
ret->convertAudioFormat = 1; ret->convertAudioFormat = 1;
if(0 != parseAudioConfig(&ret->outAudioFormat, format)) if(0 != parseAudioConfig(&ret->reqAudioFormat, format))
{ {
ERROR("error parsing format at line %i\n", ERROR("error parsing format at line %i\n",
bp->line); bp->line);
@@ -102,6 +103,8 @@ AudioOutput * newAudioOutput(ConfigParam * param) {
} }
int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) { int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) {
int ret;
if(audioOutput->open) { if(audioOutput->open) {
if(cmpAudioFormat(audioFormat, &audioOutput->inAudioFormat) if(cmpAudioFormat(audioFormat, &audioOutput->inAudioFormat)
== 0) == 0)
@@ -114,20 +117,24 @@ int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) {
copyAudioFormat(&audioOutput->inAudioFormat, audioFormat); copyAudioFormat(&audioOutput->inAudioFormat, audioFormat);
if(audioOutput->convertAudioFormat) { if(audioOutput->convertAudioFormat) {
if(cmpAudioFormat(&audioOutput->inAudioFormat, copyAudioFormat(&audioOutput->outAudioFormat,
&audioOutput->outAudioFormat) == 0) &audioOutput->reqAudioFormat);
{
audioOutput->sameInAndOutFormats = 1;
}
else audioOutput->sameInAndOutFormats = 0;
} }
else { else {
audioOutput->sameInAndOutFormats = 1;
copyAudioFormat(&audioOutput->outAudioFormat, copyAudioFormat(&audioOutput->outAudioFormat,
&audioOutput->inAudioFormat); &audioOutput->inAudioFormat);
} }
return audioOutput->openDeviceFunc(audioOutput); ret = audioOutput->openDeviceFunc(audioOutput);
if(cmpAudioFormat(&audioOutput->inAudioFormat,
&audioOutput->outAudioFormat) == 0)
{
audioOutput->sameInAndOutFormats = 1;
}
else audioOutput->sameInAndOutFormats = 0;
return ret;
} }
static void convertAudioFormat(AudioOutput * audioOutput, char ** chunkArgPtr, static void convertAudioFormat(AudioOutput * audioOutput, char ** chunkArgPtr,

1
src/audioOutput.h
View File

@@ -60,6 +60,7 @@ struct _AudioOutput {
int convertAudioFormat; int convertAudioFormat;
AudioFormat inAudioFormat; AudioFormat inAudioFormat;
AudioFormat outAudioFormat; AudioFormat outAudioFormat;
AudioFormat reqAudioFormat;
char * convBuffer; char * convBuffer;
int convBufferLen; int convBufferLen;
int sameInAndOutFormats; int sameInAndOutFormats;

14
src/audioOutputs/audioOutput_alsa.c
View File

@@ -93,6 +93,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
snd_pcm_hw_params_t * hwparams; snd_pcm_hw_params_t * hwparams;
snd_pcm_sw_params_t * swparams; snd_pcm_sw_params_t * swparams;
unsigned int sampleRate = audioFormat->sampleRate; unsigned int sampleRate = audioFormat->sampleRate;
unsigned int channels = audioFormat->channels;
snd_pcm_uframes_t alsa_buffer_size; snd_pcm_uframes_t alsa_buffer_size;
snd_pcm_uframes_t alsa_period_size; snd_pcm_uframes_t alsa_period_size;
unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME; unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME;
@@ -107,10 +108,10 @@ static int alsa_openDevice(AudioOutput * audioOutput)
bitformat = SND_PCM_FORMAT_S16; bitformat = SND_PCM_FORMAT_S16;
break; break;
case 24: case 24:
bitformat = SND_PCM_FORMAT_S16; bitformat = SND_PCM_FORMAT_S24;
break; break;
case 32: case 32:
bitformat = SND_PCM_FORMAT_S16; bitformat = SND_PCM_FORMAT_S32;
break; break;
default: default:
ERROR("Alsa device \"%s\" doesn't support %i bit audio\n", ERROR("Alsa device \"%s\" doesn't support %i bit audio\n",
@@ -162,14 +163,15 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto fail; goto fail;
} }
err = snd_pcm_hw_params_set_channels(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
audioFormat->channels); &channels);
if(err < 0) { if(err < 0) {
ERROR("Alsa device \"%s\" does not support %i channels: " ERROR("Alsa device \"%s\" does not support %i channels: "
"%s\n", ad->device, (int)audioFormat->channels, "%s\n", ad->device, (int)audioFormat->channels,
snd_strerror(-err)); snd_strerror(-err));
goto fail; goto fail;
} }
audioFormat->channels = channels;
err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
&sampleRate, 0); &sampleRate, 0);
@@ -178,6 +180,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
ad->device, (int)audioFormat->sampleRate); ad->device, (int)audioFormat->sampleRate);
goto fail; goto fail;
} }
audioFormat->sampleRate = sampleRate;
err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
&alsa_buffer_time, 0); &alsa_buffer_time, 0);
@@ -214,6 +217,9 @@ static int alsa_openDevice(AudioOutput * audioOutput)
audioOutput->open = 1; audioOutput->open = 1;
DEBUG("alsa device \"%s\" will be playing %i channel audio at %i Hz\n",
ad->device, channels, sampleRate);
return 0; return 0;
error: error: